EP1913584A2 - Appareil et procede de codage et de decodage de signal audio - Google Patents

Appareil et procede de codage et de decodage de signal audio

Info

Publication number
EP1913584A2
EP1913584A2 EP06769219A EP06769219A EP1913584A2 EP 1913584 A2 EP1913584 A2 EP 1913584A2 EP 06769219 A EP06769219 A EP 06769219A EP 06769219 A EP06769219 A EP 06769219A EP 1913584 A2 EP1913584 A2 EP 1913584A2
Authority
EP
European Patent Office
Prior art keywords
subdivision
block
information
channel
length
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Withdrawn
Application number
EP06769219A
Other languages
German (de)
English (en)
Other versions
EP1913584A4 (fr
Inventor
Tilman Liebchen
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
LG Electronics Inc
Original Assignee
LG Electronics Inc
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Priority claimed from PCT/KR2005/002290 external-priority patent/WO2007011078A1/fr
Priority claimed from PCT/KR2005/002291 external-priority patent/WO2007011079A1/fr
Priority claimed from PCT/KR2005/002292 external-priority patent/WO2007011080A1/fr
Priority claimed from PCT/KR2005/002306 external-priority patent/WO2007011083A1/fr
Priority claimed from PCT/KR2005/002307 external-priority patent/WO2007011084A1/fr
Priority claimed from PCT/KR2005/002308 external-priority patent/WO2007011085A1/fr
Application filed by LG Electronics Inc filed Critical LG Electronics Inc
Publication of EP1913584A2 publication Critical patent/EP1913584A2/fr
Publication of EP1913584A4 publication Critical patent/EP1913584A4/fr
Withdrawn legal-status Critical Current

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Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/0017—Lossless audio signal coding; Perfect reconstruction of coded audio signal by transmission of coding error
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/0212—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using orthogonal transformation
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/022—Blocking, i.e. grouping of samples in time; Choice of analysis windows; Overlap factoring
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/167—Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
    • G—PHYSICS
    • G11—INFORMATION STORAGE
    • G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
    • G11B27/00—Editing; Indexing; Addressing; Timing or synchronising; Monitoring; Measuring tape travel
    • G11B27/10—Indexing; Addressing; Timing or synchronising; Measuring tape travel
    • G11B27/102—Programmed access in sequence to addressed parts of tracks of operating record carriers
    • G11B27/105—Programmed access in sequence to addressed parts of tracks of operating record carriers of operating discs
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04S—STEREOPHONIC SYSTEMS 
    • H04S1/00—Two-channel systems
    • H04S1/007—Two-channel systems in which the audio signals are in digital form
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/032—Quantisation or dequantisation of spectral components
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/16—Vocoder architecture
    • G10L19/18—Vocoders using multiple modes
    • G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
    • G—PHYSICS
    • G11—INFORMATION STORAGE
    • G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
    • G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
    • G11B20/10—Digital recording or reproducing
    • G11B20/10527—Audio or video recording; Data buffering arrangements
    • G11B2020/10537—Audio or video recording
    • G11B2020/10546—Audio or video recording specifically adapted for audio data

Definitions

  • the present invention relates to a method for processing audio signal, and more particularly to a method and apparatus of
  • MPEG MP3 or AAC
  • DVD audio and Super CD Audio include proprietary
  • Lossless audio coding permits the compression of digital audio data without any loss in quality
  • the present invention relates to method of processing an audio
  • a channel in a frame of the audio signal is subdivided into a plurality of blocks according to a subdivision hierarchy.
  • the subdivision hierarchy has more than
  • each level being associated with a different block length. At least two of the blocks have different
  • First subdivision information may be generated indicating a number of levels in the subdivision hierarchy
  • second subdivision information may be generated indicating
  • the second subdivision information depends on the first
  • the first subdivision information may be
  • the audio signal included in a header of the audio signal.
  • the audio signal includes a header of the audio signal.
  • first subdivision information may include 2 bits.
  • the first subdivision information indicates
  • first subdivision information is 01, the first subdivision
  • the information indicates the subdivision hierarchy includes up to three levels; if the first subdivision information is 10, the
  • first subdivision information indicates the subdivision hierarchy includes four levels; and if the first subdivision information is 11, the first subdivision information indicates
  • the subdivision hierarchy includes five levels.
  • subdivision hierarchy includes five levels.
  • the second subdivision information is included in a header of the audio signal.
  • the second subdivision information may be
  • each information bit is associated with a level
  • each information bit indicates
  • the method further includes predicting current data samples in the channel based on previous data
  • the present invention further relates to methods and
  • FIG. 1 is an example illustration of an encoder according to
  • FIG. 2 is an example illustration of a decoder according to an
  • FIG. 3 is an example illustration of a bitstream structure
  • FIG. 4 is an example illustration of a conceptual view of a
  • FIG. 5 is an example illustration of a block switching examples and corresponding block switching information codes.
  • FIG. 6 is an example illustration of block switching methods
  • FIG. 1 is an example illustration of an encoder 1 according to
  • a partitioning part 100 partitions the input audio data into
  • each channel may be further
  • a buffer 110 stores block and/or frame samples partitioned by
  • a coefficient estimating part 120 estimates an optimum set of
  • the order of the predictor can be adaptively chosen as
  • the coefficient estimating part 120 calculates a set of parcor values for the block of digital audio data.
  • a quantizing part 130 quantizes the set of parcor values .
  • a first entropy coding part 140 calculates parcor residual values by subtracting an offset value from the parcor value
  • the entropy parameters are chosen from an optimal table.
  • optimal table is selected from a plurality of tables based on
  • plurality of tables are predefined for a plurality of sampling
  • a coefficient converting part 150 converts the quantized
  • a predictor 160 estimates current prediction values from the previous original samples stored in the buffer 110 using the
  • a second entropy coding part 180 codes the prediction residual
  • the second entropy coding part 180 may code the prediction residual using one of two alternative coding
  • BGMC Block Gilbert-Moore Codes
  • a multiplexing part 190 multiplexes coded prediction residual, code indices, coded parcor residual values, and other additional information to form a compressed bitstream.
  • the encoder 1 also provides a cyclic redundancy check (CRC)
  • the CRC can be used to checksum, which is supplied mainly for the decoder to verify the decoded data.
  • the CRC can be used to
  • Additional encoding options include flexible block switching
  • the encoder 1 may use these options to offer several compression levels with different complexities.
  • the joint channel coding * is used to
  • FIG. 2 is an example illustration of a decoder 2 according to
  • FIG. 2 shows the lossless audio signal decoder which is significantly less
  • a demultiplexing part 200 receives an audio signal and
  • a first entropy decoding part 210 decodes the parcor residual values using entropy
  • a second entropy decoding part 220 decodes the demultiplexed coded prediction residual using
  • a coefficient converting part 230 converts
  • predictor 240 estimates a prediction residual of the block of
  • An adder 250 adds the decoded prediction residual to the estimated
  • An assembling part 260 assembles the decoded block data into frame data.
  • the decoder 2 decodes the coded prediction residual and the parcor residual values, converts the parcor residual
  • FIG. 3 is an example illustration of a bitstream structure of a compressed audio signal including a plurality of channels (e.g., M channels) according to the present invention.
  • the bitstream consists of at least one audio frame including a
  • Each channel is sub ⁇
  • Each sub-divided block has a different size and includes coding data according to the encoding of FIG.l.
  • the coding data within a subdivided block has a different size and includes coding data according to the encoding of FIG.l.
  • partition is identical for both channels, and blocks are
  • bitstream configuration syntax (Table 6) indicates whether
  • the block partition for each channel is independent .
  • An aspect of the present invention relates to subdividing each channel into a plurality of blocks prior to using the actual
  • FIG. 4 is an example illustration of a conceptual view of a
  • FIG. 4 illustrates a method of
  • each channel is provided in a single frame, each
  • channel may be subdivided (or partitioned) to up to 32 blocks,
  • the block switching method according to the present invention is performed by the partitioning part 100
  • channel of N samples is either encoded using one full length
  • N B N
  • N B N
  • N/4 e.g. 1:4 switching
  • this scheme may have some limitations. For example, while only 1:1 or 1:4 switching may be used.
  • each channel of a frame may be hierarchically subdivided into a plurality of blocks.
  • FIG. 4 illustrates a channel which can be
  • N/32 may be possible within a channel according to the
  • each block results from a subdivision of a superordinate block of double length.
  • partition into N/4 + N/2 + N/4 may not be possible (e.g.,
  • each block has a length equal to one of ,
  • N is the length of the channel
  • m is an integer greater than or equal to 2
  • p represents a number of the levels in the subdivision hierarchy.
  • a bitstream includes information indicating block switching
  • settings are made so that a minimum block size
  • N B N/16, this indicates that the block switching process has been hierarchically performed 4 times, which is referred to as a level 4 block switching.
  • level 0 block switching which is referred to as a level 0 block switching.
  • the information indicating the block switching level will be referred to as a first block switching information.
  • the first block switching information For example, the first
  • block switching information may be represented by a 2-bit
  • the second block switching information may be
  • one channel configures one block.
  • Table 1 Block switching levels.
  • the bs_info field may include up to 4 bytes in accordance with
  • mapping of bits with respect to levels 1 to 5 may be [(0)1223333 44444444 55555555
  • the first bit may be reserved for indicating independent or synchronous block switching, which is described in more detail below in the Independent/Synchronous Block
  • FIGs. 5 (a) -5 (f) illustrate different block
  • N B N/8, and the bs_info consists of one byte.
  • N B N, the bits of
  • bs_info are set if a block is further subdivided. For example,
  • FIG. 5(f) could not have been obtained by subdividing a block of the previous level .
  • FIGs. 6 (a) - 6(c) are example illustrations of block switching
  • FIG. 6 (a) illustrates an example where
  • FIG. 6 (b) illustrates an example in which two channels
  • channels 1 and 2 configure one channel pair, and block
  • channels 1 and 2 is performed independently.
  • the channel pair refers to two arbitrary audio channels .
  • the decision on which channels are grouped into channel pairs can be made automatically by the encoder or manually by the user, (e.g., L and R channels, Ls and Rs channels) .
  • each channel may be identical for all channels
  • the block switching can be performed individually for each channel. Namely, as
  • the channels may be divided into blocks
  • channels of a channel pair may be block switched synchronously.
  • the channels are block switched synchronously.
  • the channels are block switched synchronously. In synchronous block switching, the channels are block
  • FIG. 1 is a diagrammatic representation of a single block switched (i.e., divided into blocks) in the same manner.
  • the blocks may be interleaved. If the two channels of a
  • the described method of independent or synchronous block switching may be applied to a multi-channel group having
  • a number of channels equal to or more than 3 channels .
  • channels of a multi-channel group are not correlated with each
  • each channel of the multi-channel group may be switched
  • the "bs_info" field is used as the information for
  • w bs_info field is also used as the information for indicating whether block switching has been performed independently or
  • a particular bit within the "bs_info" field may be used. If, for example, the two channels of the channel pair are
  • the first bit of the "bs_info" field is set as "0".
  • FIGs. 6 (a) , 6 (b) , and 6(c) will now be described in detail. Referring to FIG. 6 (a) , since none of the channels perform
  • channels 1 and 2 configure a channel pair, wherein the two channels are synchronous to one another
  • a block of one channel e.g., a block of one channel
  • channels 1 and 2 configure a channel
  • channel 1 is split into
  • channel data may be arranged separately.
  • Joint channel coding also called joint stereo, can be used to
  • the channels can be rearranged by
  • the encoder in order to assign suitable channel pairs.
  • lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also Besides simple difference coding, lossless audio codec also
  • the present invention relates to audio lossless coding and is
  • Random access stands for fast access to any part of the encoded audio signal without costly
  • the encoder needs to insert a frame that
  • the inserted frame is referred to as a "random access frame" .
  • random access frame In such a
  • a "random_access" field is used as
  • random access is not supported. In other words, when w random_access > 0", random access is supported. More
  • the distance is expressed by the number of frames.
  • configuration syntax (Table 6) may further include information
  • This information is represented as a 2-bit "ra_flag" field. More specifically, for example,
  • the "random_access" field within the configuration syntax may also be referred to as
  • the "ra_flag" field may also be used to indicate the "ra_flag" field.
  • an audio signal includes
  • each random access unit containing one or more audio data frames, one of which is a random access frame, wherein
  • the configuration information includes first general
  • the random access unit size information indicating a distance between two adjacent random access unit
  • method of decoding an audio signal includes receiving the
  • each random access unit containing one
  • the first general information indicating a distance between two adjacent random access
  • the configuration information the second general information indicating where random access size information for each random access unit is stored, and the random access unit size information indicating a distance between two adjacent random access frames in bytes .
  • an audio signal includes multi-channels
  • each channel may be mapped at a one-to-one correspondence with
  • the "chan_config__info” field includes information for mapping the channels to the loudspeaker
  • channels field is equal to "0", this indicates that the channel corresponds to a mono channel.
  • present invention also includes information indicating whether
  • Table 2 Channel configuration.
  • an audio signal includes multiple or
  • samples a 32-bit "samples" field is used as information indicating the total number of audio data samples configuring each channel.
  • a 16-bit "frame_length” field is used as information indicating the number of samples for each channel within the corresponding frame .
  • a 16-bit value of the "frame__length” field is determined by a value used by the encoder, and is referred to as a user-defined value. In other words, instead of being a fixed value, the user-defined value is arbitrarily determined upon the encoding process . Therefore, during the decoding process, when the bitstream is received through the demultiplexing part 200 of shown in FIG. 2, the frame number of each channel should first be obtained. This value is obtained according to the algorithm shown below.
  • the total number of frames for each channel is calculated by dividing the total number of samples for each
  • samples field is an exact multiple of the number of samples within each frame, which is decided by the "frame_length” field, the multiple value becomes the total number of frames. However, if the total number of samples decided by the
  • samples field is not an exact multiple of the number of
  • the encoder may freely decide and transmit the total number of samples ("samples" field) for
  • the decoder may
  • linear prediction is applied for the lossless audio coding.
  • the predictor 160 shown in FIG. 1 includes at least one or more filter coefficients so as to predict a current sample value from a previous sample value.
  • the second entropy coding part 180 performs entropy coding on a residual value corresponding to the difference between the predicted value and the original value .
  • the predictor coefficient values for each block that are applied to the predictor 160 are selected as optimum values from the coefficient estimating part 120.
  • the predictor coefficient values are entropy coded by the first entropy coding part 140.
  • the data coded by the first entropy coding part and the second entropy coding part 180 are inserted as part of the bitstream by the multiplexing part 190 and then transmitted.
  • Linear prediction is used in many applications for speech and audio signal processing.
  • an exemplary operation of the predictor 160 will be described based on Finite Impulse Response (FIR) filters.
  • FIR Finite Impulse Response
  • the current sample of a time-discrete signal x(ri) can be
  • JC is the order of the predictor.
  • the backward adaptation procedure has
  • backward-adaptive codecs have the drawback that the adaptation has to be carried out both in the encoder and the decoder, making the decoder significantly more complex than in the forward-adaptive case.
  • forward adaptive prediction will be given as an example in the
  • a minimized variance of the residual are usually estimated for each block by the coefficient estimating part 120 using the autocorrelation method or the covariance method.
  • autocorrelation method using the conventional Levinson-Durbin algorithm, has the additional advantage of providing a simple means to iteratively adapt the order of the predictor.
  • the algorithm inherently calculates the corresponding parcor coefficients as well.
  • Another aspect of forward-adaptive prediction is to determine a suitable prediction order. Increasing the order decreases
  • bit rate R e for the residual.
  • bit rate R e for the residual.
  • the residual can be derived, resulting in an estimate of the expected bit rate for the residual. Together with the bit rate for the coefficients, the total bit rate can be determined in each iteration, i.e., for each prediction order. The optimum order is found at the point where the total bit rate no longer decreases .
  • the prediction order K which decides the number of predictor coefficients for linear prediction.
  • the prediction order K is also determined by the coefficient estimating part 120.
  • information on the determined prediction order is included in
  • bitstream the bitstream and then transmitted.
  • the configuration syntax (Table 6) includes information
  • 10-bit "max_order” field corresponds to information indicating a maximum order value.
  • the configuration syntax (Table 6) includes a 1-bit "adapt_order"
  • the "max_order" field becomes the final order applied to all of the blocks .
  • the optimum order (opt_order) is decided based upon the value of max_order field and the size (N B ) of the corresponding
  • each block may be decided considering the size of the corresponding block.
  • the opt_order value being
  • the present invention relates to higher prediction orders.
  • this factor can be increased (e.g., up to 32), enabling a
  • K m ⁇ 1023.
  • K ms ⁇ may be bound
  • Table 8 can also be up to 10 bits. The actual number of
  • bits in a particular block may depend on the maximum order
  • prediction order may be smaller than a global prediction order.
  • the local prediction order is determined from
  • the "opt_order” field is determined on 8 bits (instead
  • the opt_order may be determined based on
  • a channel are predicted.
  • a first sample of a current block is predicted using the last K samples of a previous block. The K
  • the first sample in the block does not perform
  • the random access frame corresponds to a reference frame of
  • the random access frame does not perform prediction by using the previous frame sample. Namely, this progressive prediction technique may be applied at the beginning of the random access frame.
  • predictor coefficients h k is not very efficient for
  • coefficient estimating part 120 As described above, for example, the coefficient estimating part 120 is processed
  • coding part 140 is used for coding the above-described
  • the related art Rice code is used as the first entropy coding method according to the present invention. For example,
  • the first entropy coding part 140 which, in turn, are encoded by using the first entropy coding part 140, e.g., the Rice code method.
  • the corresponding offsets and parameters of Rice code used in this process can be globally chosen from one of the sets shown in Table 3, 4 and 5 below.
  • a table index i.e., a 2-bit "coef__table”
  • r(.) is an empirically determined mapping table (not shown
  • the mapping table may vary with implementation) . Accordingly, the three types of coefficient tables used for the first entropy coding are provided according to the sampling frequency.
  • the sampling frequency may be any suitable sampling frequency.
  • tables can be chosen for the entire file.
  • the table should typically be chosen depending on the sampling rate. For material with 44.1 kHz, the applicant of the present invention
  • table can also be chosen by other criteria.
  • Table 3 Rice code parameters used for encoding of
  • Table 4 Rice code parameters used for encoding of
  • Table 5 Rice code parameters used for encoding of quantized coefficients (192 kHz) .
  • the present invention contains two different modes of the
  • FIG. 1 which will now be described in detail.
  • the indices of the applied codes are transmitted, as shown in FIG. 1. Since there are different ways to determine
  • the encoder can use a more complex and efficient coding scheme using BGMC mode.
  • BGMC mode In the BGMC mode,
  • the two types include residuals that belong to a central region of the distribution,
  • tails are simply re-centered (i.e., for e(n) > e max ,
  • the BGMC first splits the
  • the BGMC transmits LSBs using direct fixed-lengths
  • transmitted LSBs may be selected such that they only slightly
  • the coding to be significantly less complex.
  • the configuration syntax (Table 6) and the block_data syntax (Table 8) according to the present invention include
  • the configuration syntax (Table 6) first includes a 1-bit
  • the "sb_part" field corresponds to information related
  • the "ec_sub” field indicates the number of sub-
  • BGMC code or the BGMC code is used to partition the block to sub-
  • code is used to partition the block to sub-blocks at a 1:2:4:8 rate.
  • 2 bits are assigned to the w ec_sub" field.
  • Second entropy coding part 180 are coded by second entropy coding part 180 using a difference coding method.
  • An example of using the Rice code will now be
  • the ⁇ X ec_sub" field in the block-data syntax indicates whether one or four blocks are used.
  • bitstreara are shown in the tables below.
  • Table 6 shows a
  • configuration syntax may form a header periodically placed in
  • bitstream may form a header of each frame; etc.
  • Table 8 shows a block-data
  • Table 7 Frame_data syntax.
  • Table 8 Block data syntax.
  • the lossless audio codec is compared with two of the most popular programs for lossless audio
  • the compression ratio is defined as:
  • ALS at maximum level outperforms both FLAC and Monkey's Audio for all formats, but particularly for high-definition material (i.e., 96 kHz / 24-bit and above). Even at medium level, ALS delivers the best overall compression.
  • Table 10 Average CPU load (percentage on a 1.7 GHz Pentium-M) , depending on audio format (kHz/bits) and ALS
  • the codec is designed to offer a large range of complexity
  • the decoder may be

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Abstract

Dans un mode de réalisation de cette invention, un canal d'une trame du signal audio est divisé en plusieurs blocs en fonction d'une hiérarchie de subdivision, laquelle possède plus d'un niveau, chaque niveau étant associé à une longueur de bloc différente. Au moins deux de ces blocs ont des longueurs différentes. Une première information de subdivision indiquant un nombre de niveaux dans la hiérarchie de subdivision peut être générée et une seconde information de subdivision indiquant que le canal a été divisé en blocs peut être générée. La longueur de la seconde information de subdivision dépend la première information de subdivision.
EP06769219A 2005-07-11 2006-07-10 Appareil et procede de codage et de decodage de signal audio Withdrawn EP1913584A4 (fr)

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US69755105P 2005-07-11 2005-07-11
PCT/KR2005/002290 WO2007011078A1 (fr) 2005-07-16 2005-07-16 Appareil et procede de codage et de decodage d'un signal audio
PCT/KR2005/002291 WO2007011079A1 (fr) 2005-07-16 2005-07-16 Appareil et procede de codage et de decodage de signal audio
PCT/KR2005/002292 WO2007011080A1 (fr) 2005-07-16 2005-07-16 Appareil et procede de codage et decodage d'un signal audio
PCT/KR2005/002306 WO2007011083A1 (fr) 2005-07-18 2005-07-18 Appareil et procede de codage et de decodage d'un signal audio
PCT/KR2005/002307 WO2007011084A1 (fr) 2005-07-18 2005-07-18 Appareil et procede de codage et de decodage de signal audio
PCT/KR2005/002308 WO2007011085A1 (fr) 2005-07-18 2005-07-18 Appareil et procede destines a coder et decoder un signal audio
US70057005P 2005-07-19 2005-07-19
PCT/KR2006/002681 WO2007008003A2 (fr) 2005-07-11 2006-07-10 Appareil et procede de codage et de decodage de signal audio

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